Voice over Internet Protocol (VoIP) systems employ session control protocols, such as the Session Initiation Protocol (SIP), to control the set-up, modification, and tear-down of calls as well as the selection of audio and video codecs, which encode speech and video allowing transmission over an IP network as digital audio and video via one or more media streams. The advantage to VoIP is that a single network can be utilized to transmit data packets as well as voice and video packets between users, thereby greatly simplifying communications. SIP is an open signaling protocol for establishing many kinds of real-time and near-real-time communication sessions, which may also be referred to as dialogs. Examples of the types of communication sessions that may be established using SIP include voice, video, and/or instant messaging. These communication sessions may be carried out on any type of communication device such as a personal computer, laptop computer, telephone, cellular phone, Personal Digital Assistant, etc. One key feature of SIP is its ability to use an end-user's Address of Record (AOR) as a single unifying public address for all communications. Thus, in a world of SIP-based communications, a user's AOR becomes their single address that links the user to all of the communication devices associated with the user. Using this AOR, a caller can reach any one of the user's communication devices, also referred to as SIP User Agents (UAs), without having to know each of the unique device addresses or phone numbers.
That is, SIP supports the initiation, modification, and termination of media sessions between SIP UAs. These sessions are managed by SIP dialogs, which represent a SIP relationship between a pair of user agents. Because dialogs are between pairs of SIP UAs, SIP'S usage for two-party communications (such as a phone call) is obvious. Communications sessions with multiple participants, however, are more complicated. SIP can support many models of multi-party communications. One, referred to as loosely coupled conferences, makes use of multicast media groups. In the loosely coupled model, there is no signaling relationship between participants in the conference and there is no central point of control or conference server.
In another model, sometimes referred to as the tightly coupled conference, there is a central point of control. Each participant connects to this central point. The central point provides a variety of conference functions and may possibly perform media mixing functions as well. Tightly coupled conferences are not directly addressed by RFC 3261, although basic participation is possible without any additional protocol support. Currently, when a conference call is established between SIP User Agents, the conference is controlled by an entity that would act as a Conference Controller. The media for the conference call generally flows through one or more Media Gateways, which are controlled by the Conference Controller utilizing a media control protocol like H.248/Megaco or MSML. If the Conference Controller restarts or experiences a service interruption due to some event, such as a power failure or other catastrophic event, the media path between one or more of the SIP User Agents and the Media Gateway may be maintained, but the call dialog, e.g. signaling paths, is never reestablished. That is, the call may stay in a connection preservation mode where the media is available but the signal pathways are not. Accordingly, features such as adding a new participant, transferring calls, and placing a call on hold for example, will not work using the existing media path while in the connection preservation mode.